I am running an Asterisk server (which has gone from 1.x to 1.2b2 at the
moment) that has 3 X100P cards and around 10 SIP phones in my office and I
have a problem when I want to redirect my desk phone to my cell phone.
I have a Polycom 600 phone on my desk (I have also tried this with Aastra
and Grandstream phones). If I choose the forward option and enter my cell
number, the next call will ring my cell but I will get no audio on my side
most of the time. After the call ends both incoming and outgoing Zap
interfaces will report that the call is still on and will continue that way
until I manually destroy the channel. Why doesn't either interface detect
the
end of the call? The result is that I cannot forward my phone because 2 lines
will be engaged untill manually reset.
--
Carlos Chavez
Director de Tecnolog?a
Telecomunicaciones Abiertas de M?xico S.A. de C.V.
Tel: +52-55-91169161 Ext 2001